similar to: Newbie with IAX2

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Hello all, I have a WildCard TE410P setup and working with a full PRI with the latest CVS. SIP and IAX2 gateways are accessing the PRI without issue, however; echo is very prominent in some calls and is only heard by the IAX2/SIP client. The echo Is not present in calls to cell phones because they are digital, centrex land Lines have a barely noticable echo, but analog lines aint so pretty.

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Is it possible to allow distinctive rings work for FXS ports as well? I need a certain FXS extension to ring a distinctive double ring. I modified zapata.conf appropriately for dring1,dring2 and it just Seems to ignore my updates. Do distinctive rings only work for FXO ports? Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf

Hello all, I have a TE410P hooked up to a single PRI. Incomming CallerID is fine, and Outgoing works as well. However, if I change my dialplan for an extension To do a 'follow you, follow me' or setup an auto attendant that rings extensions Thru to cell phones, the CallerID always shows up on the call reciever as '708'. When I do a verbose dump at the console, it appears that

Hello all, I updated from CVS a few days ago and noticed that my calls just cut out without reason. The CLI says this: -- Hungup 'Zap/3-1' It occurs without error or warning. Is their a bug in CVS asterisk or libpri? This never occurred before. Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf Type:

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

Hi all - We have several servers working just fine with IAX2 w/o trunk mode. We are trying to setup trunk mode to save bandwidth, but we can not achieve the savings with our current configuration (see below). When we place a call between * boxes A & B it works fine, but the command 'iax2 trunk debug' shows no activity for the trunk mode (1 peer, 0 calls). Anyone who has

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

I have 2 Asterisk servers connected via IAX2 connections. PBX1 is on the internet with a public IP Address - with PRI PBX 2 is behind a NAT router with IAX2 Ports forwarded 1-way audio is an issue with incoming and outgoing calls using the PRI. However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there are no issues. As well as a restart of asterisk on PBX2

Phone - TDM430P - home* - IAX2 - office* - PRI - Telco I dial a busy number from the Phone. Home* shows this in the CLI: -- Executing Macro("Zap/1-1", "dial-wu|2922004") in new stack -- Executing Dial("Zap/1-1", "IAX2/andrew@wu-ast/2922004||g") in new stack -- Called andrew@wu-ast/2922004 -- Call accepted by wu-ast (format gsm) --

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Hello ! Asterisk 1.0.9 running on Linux 2.6.12. I'm not able to call iax2 channels. There can be no translation path found. When I try to call from a ZAP PRI channel the following error occurs: channel.c:1891 ast_request: No translator path exists for channel type IAX2 (native 63488) to 72 dial_exec: Unable to create channel of type 'IAX2' What is wrong ? Here is my iax.conf:

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Hi, I'm using IAX2 between our SIP and PSTN servers, both running Asterisk 1.6.2. Users connect to the SIP server and dial; the SIP server forwards the call to the PSTN server over IAX2, which then dials out over the connected PRI. Since users need detailed call progress feedback, the first action in the dialplan on the PSTN server side is Answer(). In this scenario it's easy for

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to